Digital cellular telephone networks have traditionally relied upon circuit switched channels to carry user traffic such as voice communications A circuit switched channel is formed by the allocation of one slot per frame in a given TDMA channel. Whilst circuit switched sessions have proved adequate for voice calls, they do not provide an efficient mechanism for transferring large amounts of data which is “bursty” in nature. For example, the setting up of a circuit switched session to download a web page from a web server is likely to result in the connection remaining idle for significant amounts of time, and being overloaded when there is data to transmit.
To facilitate fast data transfers to mobile terminals, packet switched data services are being introduced to digital cellular telephone networks. For example, the General Packet Radio Service (GPRS) is currently being introduced to many GSM networks. Unlike circuit switched calls, a GPRS session (referred to as a PDP context) for a given user does not necessarily occupy one slot per frame on a given TDMA channel. Rather, slots are only used when the user has data to send or receive. When there is no traffic to transmit, no slots are allocated to the user. When there is a large volume of data to transmit, the user may be allocated one or more slots per frame.
GPRS will be available in future third generation networks such as 3G networks which will rely upon CDMA rather than TDMA. 3G networks will however continue to provide circuit switched services at least for the foreseeable future, although these sessions will not necessarily be end to end. Rather, the links between mobile terminals and the networks will be circuit switched, with data being routed within and between networks via high capacity packet switched networks (which have sufficient bandwidth to handle real time traffic).
It is envisaged that in the future, the packet switched (access) domain will be able to carry real time information streams, for example relating to voice and video telephony. However, at present the transmission reliability of GPRS is not sufficient to provide users with telephony services of the quality which they will expect, hence the continued provision of circuit switched services (the provision of circuit switched services is also likely to be necessary by the need to continue to service older mobile terminal equipment which is not GPRS enabled).
To facilitate the provision of multimedia services via the packet switched “domain”, the 3rd Generation Partnership project (3GPP) responsible for the 3G standards has been developing a so-called IP Multimedia Core Network Subsystem (IMS). IMS communicates with the GPRS core network and contains all elements that are used to provide IP based multimedia services. The base protocol for multimedia services is the IETF Session Initiation Protocol (SIP). SIP makes it possible for a calling party to establish a packet switched session to a called party (using so-called SIP User Agents, UAs, installed in the user terminals) even though the calling party does not know the current IP address of the called party prior to initiating the call. SIP provides other functionality including the negotiation of session parameters (e.g. Quality of Service and codecs).
FIG. 1 illustrates schematically a 3G network providing circuit switched (CS) and packet switched (PS) access networks to a mobile terminal. The figure illustrates a call being made by the mobile terminal, via its circuit switched access network, to a PC which has access only to a packet switched network. The session is initiated by the dialling of a telephone number from the mobile terminal, i.e. this does not involve any exchange of SIP signalling between the home network and the mobile terminal, and SIP URLs cannot be transferred over the CS domain. The destination terminal must have allocated to it a standard telephone number in order for such a session to be established. Translation between circuit switched and packet switched data is performed by an interworking gateway (GW), with the GW establishing the packet switched session to the PC using SIP signalling. Assuming that the packet switched network used by the PC has sufficient bandwidth (e.g. the network is a broadband network), the call will provide the users with a sufficient level of quality for voice and video. In this scenario, the IMS of the home operator's network is not used.
In addition to the need for the destination terminal to have allocated to it a telephone number, a further disadvantage of the architecture of FIG. 1 is that the destination terminal will not necessarily know that a conversational bearer has been established using a CS access network. Any attempt by the destination terminal to establish some additional (non-conversational) PS bearer will fail, because the gateway cannot provide this service. Also, any attempt by the initiating terminal to establish a (non-conversational) PS bearer may fail because the destination terminal will not be able to associate the set-up request with the existing conversational bearer.
FIG. 2 illustrates an alternative scenario in which a call between the mobile terminal and the PC is established using the PS access network available to the mobile terminal. The call is established using a SIP server of the IMS. Due to the limited bandwidth of the PS access network available to the mobile terminal, the session is unlikely to be of sufficient quality to handle real time voice and video data. A separate CS bearer should be established for this purpose. However, this might not be straightforward given that the initiating or terminating terminal might know only the SIP URL of the peer terminal, and not its telephone number.
It is likely that users will prefer to initiate and receive circuit switched and packet switched calls using a common signalling interface. However, under the current proposals, a user would initiate and receive a packet switched call using SIP, e.g. to initiate a packet switched call the user would enter the SIP address for the called party (e.g. john@example.org), whilst he/she would initiate and receive a circuit switched call using the DTAP protocol, e.g. to initiate such a call the user would dial the called party's telephone number (e.g. 012345 . . . ). Network operators would also prefer to use a common signalling interface as this will ease the migration of circuit switched services to the packet switched domain, when that domain has evolved sufficiently to provide the required services.